app_queue: Make queue reload members and variants of that work
Prior to this patch, 'queue reload members' cli command did not
work at all. This also affects the manager function 'QueueReload'
when supplied with the 'members: yes' field.
(closes issue AST-956)
Reported by: John Bigelow
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Merged revisions 373298 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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When processing RFC 2833 DTMF, accomodate increasing timestamps in End events
While endpoints should not be changing the source timestamp between DTMF event
packets, the fact is there exists those endpoints that do exactly that. To
work around this, we absorb timestamps within the expected re-transmit period.
Note that this period only affects End of Event packets, so it should not
prevent the detection of new DTMF digits that happen to arrive right on top
of each other.
(closes issue ASTERISK-20424)
Reported by: Vladimir Mikhelson
Tested by: mjordan, Vladimir Mikhelson
Fix incorrect MeetME conference bridge reference count decrementing and sometimes premature destruction.
When using the 'e' or 'E' option to MeetMe the configured conference bridges are loaded and examined to see
if any are empty. If no conference bridges are empty the caller is prompted to enter the number of one.
This operation left around a pointer to the last created conference bridge still containing participants.
When the caller that was not able to find any empty conference bridge hung up this pointer was disposed of
and the reference count of the conference bridge decremented. If there was only a single participant in the
conference bridge it was ultimately destroyed prematurely.
(closes issue AST-994)
Reported by: John Bigelow
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Merged revisions 373242 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Ensure that all ConfBridge sounds can be set using CONFBRIDGE function
The CONFBRIDGE function can be used to set the sounds in a ConfBridge
bridge profile. Unfortunately, three sounds were missed in the portion
of the code that applies the settings passed in from the function:
sound_only_one, join, and leave. This patch makes sure that the sounds
passed from the function are applied to the bridge profile.
Fix a regression where direct media was not permitted for calls using SIP INFO DTMF.
A change was committed to fix direct media ACL support. This change wrongly assumed that
only a single channel technology structure exists for chan_sip. This is in fact false as
a second exists for calls using SIP INFO DTMF. The code which performs direct media ACL
checking now checks for both the non-INFO DTMF and INFO DTMF channel technology structures.
Don't crash when passing a NULL message to __astman_get_header.
Before this commit, __astman_get_header would blindly dereference the passed in
'struct message *' to traverse the header list. There are cases, however, such
as '*CLI> sip qualify peer foo' where the message pointer is NULL, so we need
to check for that.
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Merged revisions 373131 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Made companding law for SS7 calls only determined by SS7 signaling type.
For SS7, the companding law for a call was chosen inconsistently depending
upon ss7type (ITU vs ANSI) and the DAHDI companding default (T1 vs E1).
For incoming calls, the companding law was determined by ss7type. For
outgoing calls, the companding law was determined by the DAHDI default.
With the wrong combination you would get A-law/u-law conflicts. An
A-law/u-law conflict sounds like bad static on the line.
SS7 ITU signaling with E1 line: ok
SS7 ITU signaling with T1 line: noise
SS7 ANSI signaling with E1 line: noise
SS7 ANSI signaling with T1 line: ok
* Fix the companding law used to be determined by the SS7 signaling type
only.
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Merged revisions 373090 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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When REF_DEBUG is enabled in certain files - most notably ccss.c - the 'tag'
parameter passed to __ao2_ref_debug will be a const char *. The function
currently expects that parameter to not be const. This causes a warning
when compiling, as the const qualifier is being discarded. With dev-mode
enabled, this prevents compiling Asterisk.
This patch makes __ao2_ref_debug's tag and file parameters const.
Resolve memory leaks in TLS initialization and TLS client connections
This patch resolves two sources of memory leaks when using TLS in Asterisk:
1) It removes improper initialization (and multiple re-initializations) of
portions of the SSL library. Asterisk calls SSL_library_init and
SSL_load_error_strings during SSL initialization; collectively this
obviates the need for calling any of the following during initialization
or client connection handling:
* ERR_load_crypto_strings (handled by SSL_load_error_strings)
* OpenSSL_add_all_algorithms (synonym for SSL_library_init)
* SSLeay_add_ssl_algorithms (synonym for SSL_library_init)
2) Failure to completely clean up all memory allocated by Asterisk and by
the SSL library for TLS clients. This included not freeing the SSL_CTX
object in the SIP channel driver, as well as not clearing the error
stack when the TLS client exited.
Note that these memory leaks were found by Thomas Arimont, and this patch
was essentially written by him with some minor tweaks.
(closes issue AST-889)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
patches:
(bugAST-889.patch) by Thomas Arimont (license 5525)
ast_waitfordigit_full would simply pass its timeout to ast_waitfor_nandfds,
expecting it to decrement the timeout by however many milliseconds were
waited. This is a problem if it consistently waits less than 1ms. The timeout
will never be decremented, and we wait... FOREVER!
This patch makes ast_waitfordigit_full manage the timeout itself. It maintains
the previously undocumented behavior that negative timeouts wait forever.
(closes issue ASTERISK-20375)
Reported by: Mark Michelson
Tested by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/2109/
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Merged revisions 373024 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Add channel name to a warning to make debugging easier.
The "autodestruct with owner in place" message is typically
indicative of a channel reference leak. Printing out the name
of the channel in the message may be helpful when trying to
debug the issue.
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Merged revisions 372932 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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chan_local: Switch from using a random 4 digit hex identifier to unique id
Changes chan_local channels to use an 8 digit hex identifier generated
atomically and sequentially in order to eliminate the chance of having
multiple channels with the same name during high call volume situations.
(issue ASTERISK-20318)
Reported by: Dan Cropp
Review: https://reviewboard.asterisk.org/r/2104/
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Merged revisions 372902 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Fix inability to shutdown gracefully due to an unending channel reference.
message.c makes use of a special message queue channel that exists
in thread storage. This channel never goes away due to the fact that
the taskprocessor used by message.c does not get shut down, meaning
that it never ends the thread that stores the channel.
This patch fixes the problem by shutting down the taskprocessor when
Asterisk is shut down. In addition, the thread storage has a destructor
that will release the channel reference when the taskprocessor is destroyed.
(closes issue AST-937)
Reported by Jason Parker
Patches:
AST-937.patch uploaded by Mark Michelson (License #5049)
Tested by Jason Parker
........
When channels get bridged due to an AMI bridge action
or a DTMF attended transfer, the two channels that
get bridged have their application data pointing to
the other channel's name. This means that if one channel
is hung up but the other moves on, it means that the
channel that moves on will have its application data
pointing at freed memory.
(issue ASTERISK-20335)
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Merged revisions 372840 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Ensure iax2 debug output is displayed when expected
When IAX2 debug was changed from iax_showframe to iax_outputframe,
some instances were missed (or added afterward). This was causing
debug output to not be displayed when expected.
(closes issue ASTERISK-20338) Reported-by: John Covert Patch-by: John Covert
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Merged revisions 372804 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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This adds a CLI warning when a SDP offer is rejected due to UDPTL
initialization failure. Previously, there was no indication of the
reason for offer rejection in this case.
(closes issue ASTERISK-20357) Reported-by: Francesco Usseglio Gaudi
........
Masquerade: Retain parkinglot settings made by CHANNEL function.
Prior to this patch, the user would have a parkinglot set on a channel that
was parked and when the channel was retrieved, any attempt by that channel
to park would simply use the default. This patch makes parkinglot values
set in this way be retained through the masquerade.
(closes issue AST-990)
Reported by: Nick Huskinson
Patches:
masquerade_parkinglot_patch.diff Uploaded by Jonathan Rose (license 6182)
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Merged revisions 372736 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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In r356604, SRTP handling was fixed to accomodate multiple crypto keys in an
SDP offer and the ability to re-create an SRTP session when the crypto keys
changed. In certain circumstances - most notably when a phone is put on
hold after having been bridged for a significant amount of time - the act
of re-creating the SRTP session causes problems for certain models of phones.
The patch committed in r356604 always re-created the SRTP session regardless
of whether or not the cryptographic keys changed. Since this is technically
not necessary, this patch modifies the behavior to only re-create the SRTP
session if Asterisk detects that the remote key has changed. This allows
models of phones that do not handle the SRTP session changing to continue
to work, while also providing the behavior needed for those phones that do
re-negotiate cryptographic keys.
Add OPENSSL_INCLUDE to the CFLAGS for ssl.c and tcptls.c.
Without this flag, those files will compile with the system installed
OpenSSL headers (if they exist). This is a real bummer if a different
path was specified using --with-ssl=
(closes issue ASTERISK-20392)
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Merged revisions 372682 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Fix VoicemailUserEntry event headers ServerEmail and MailCommand reported values.
The AMI action VoicemailUsersList VoicemailUserEntry event headers
ServerEmail and MailCommand did not report the global values if they were
not overridden. The VoicemailUserEntry event header ServerEmail was not
populated with the global value if the voicemail user did not override it.
The VoicemailUserEntry event header MailCommand was never populated with a
value.
* Removed unused struct ast_vm_user member mailcmd[].
Free ast_str objects when temp file fails to be created in MiniVM
The previous commit (r372554) was from a patch that was written before
r366880, which ensured that ast_str objects allocated in the sendmail
routine were free'd in off nominal paths. This commit frees the
string objects in the off nominal path introduced in r372554.
Fix file descriptor leak and pointer scope issue in MiniVM when sending mail
When MiniVM sends an e-mail and it has the volgain option set, it will spawn
sox in a separate process to handle the manipulation of the sound file. In
doing so, it creates a temporary file. There are two problems here:
1) The file descriptor returned from mkstemp is leaked
2) The finalfilename character pointer points to a buffer that loses scope
once volgain processing is finished.
Note that in r316265, Russell fixed some gcc warnings by using the return
value of the mkstemp call. A warning was placed in minivm that the file
descriptor was going to be leaked. This patch reverts that change, as it
handles the leak and 'uses' the file descriptor returned from mkstemp.
Ensure listed queues are not offered for completion
When using tab-completion for the list of queues on "queue reset stats"
or "queue reload {all|members|parameters|rules}", the tab-completion
listing for further queues erroneously listed queues that had already
been added to the list. The tab-completion listing now only displays
queues that are not already in the list.
(closes issue AST-963) Reported-by: John Bigelow
........
Fix loss of MOH on an ISDN channel when parking a call for the second time.
Using the AMI redirect action to take an ISDN call out of a parking lot
causes the MOH state to get confused. The redirect action does not take
the call off of hold. When the call is subsequently parked again, the
call no longer hears MOH.
* Make chan_dahdi/sig_pri restart MOH on repeated AST_CONTROL_HOLD frames
if it is already in a state where it is supposed to be sending MOH. The
MOH may have been stopped by other means. (Such as killing the generator.)
This simple fix is done rather than making the AMI redirect action post an
AST_CONTROL_UNHOLD unconditionally when it redirects a channel and thus
potentially breaking something with an unexpected AST_CONTROL_UNHOLD.
Prior to 1.8, it was not necessary for an explicit "type" to be set for an
asterisk LDAP realtime peer. Now the routine find_peer actually checks the
type field during registration and fails to find the peer if it is not set.
The attached patches make the realtime type equal whatever type is being
searched for if the type is 0 upon return from routine build_peer.
(closes issue ASTERISK-17222)
Reported by: John Covert
Patch by: David Vossel
Tested by: Darren Sessions
chan_sip: Note change in behavior to how directmediapermit/deny ACL works
r366547 introduced a change to the directmedia ACL for chan_sip which
modified the behavior significantly. Prior to the patch, this option would
bridge peers with directmedia if a peer's IP address matched its own
directmedia ACL. After that patch, the peer would check the bridged peer's
ACL instead. This change has been present since 1.8.14.0. That patched failed
to document the change in Upgrade.txt, so this patch adds mention of that
change to UPGRADE.txt (UPGRADE-1.8.txt in newer branches)
(issue AST-876)
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Merged revisions 372471 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Ensure "rules" is tab-completable for "queue show"
Previously, tabbing at the end of "queue show" produced a list of
available queues about which information could be shown, but did not
include an alternative command, "rules", to access information about
queue rules. The "rules" item should now be shown in the list of
tab-completable items.
(closes issue AST-958) Reported-by: John Bigelow
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Merged revisions 372444 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Fix DUNDi message routing bug when neighboring peer is unreachable
Consider a scenario where DUNDi peer PBX1 has two peers that are its neighbors,
PBX2 and PBX3, and where PBX2 and PBX3 are also neighbors. If the connection
is temporarily broken between PBX1 and PBX3, PBX1 should not include PBX3 in
the list of peers it sends to PBX2 in a DPDISCOVER message, as it cannot send
messages to PBX3. If it does, PBX2 will assume that PBX3 already received the
message and fail to forward the message on to PBX3 itself. This patch fixes
this by only including peers in a DPDISCOVER message that are reachable by the
sending node. This includes all peers with an empty address
(00:00:00:00:00:00) and that are have been reached by a qualify message.
This patch also prevents attempting to qualify a dynamic peer with an empty
address until that peer registers.
(closes issue ASTERISK-19309)
Reported by: Peter Racz
patches:
dundi_routing.patch uploaded by Peter Racz (license 6290)
The patch uploaded by Peter was modified slightly for this commit.
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Merged revisions 372417 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Allow configured numbers for FollowMe to be greater than 90 characters
When parsing a 'number' defined in followme.conf, FollowMe previously parsed
the number in the configuration file into a buffer with a length of 90
characters. This can artificially limit some parallel dial scenarios. This
patch allows for numbers of any length to be defined in the configuration
file.
Note that Clod Patry originally wrote a patch to fix this problem and received
a Ship It! on the JIRA issue. The patch originally expanded the buffer to 256
characters. Instead, the patch being committed duplicates the string in the
config file on the stack before parsing it for consumption by the application.
Ensure counts generated in manager_show_dialplan_helper are correct
When manager_show_dialplan_helper was written, the counter increment
for the total number of contexts was placed with the extensions
increment instead of in the enclosing loop. This function should
now generate correct context counts.
(closes issue AST-970) Reported-by: John Bigelow
........
The documentation incorrectly listed 'rtp' as a reloadable subsystem
and left out many other reloadable subsystems. It is now also
documented that subsystems may only be reloaded, not loaded or
unloaded.
(closes issue AST-977) Reported-by: John Bigelow
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Merged revisions 372354 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Fix memory leaks in app_voicemail when using IMAP storage or realtime config
This patch fixes two memory leaks:
1. When find_user is called with NULL as its first parameter, the voicemail
user returned is allocated on the heap. The inboxcount2 function uses
find_user in such a fashion when counting new messages, and fails to free
the resulting voicemail user object.
2. When populate_defaults is called on a voicemail user, it wipes whatever
flags have been set on the object by copying over the global flags object.
If the VM_ALLOCED flag was ste on the voicemail user prior to doing so,
that flag is removed. This leaks the voicemail user when free_user is later
called.
(closes issue ASTERISK-19155)
Reported by: Filip Jenicek
patches:
asterisk.patch2 uploaded by Filip Jenicek (license 6277)
dsp.c: Fix multiple issues when no-interdigit delay is present, and fast DTMF 50ms/50ms
Revert DTMF hit/miss detector to original -r349249 method with some changes, remove unnecessary;
1. reseting of hits=0, when no signal, only need to set it once.
2. incrementing of hits, when the hit is the same as the current hit.
3. setting of lasthit, when it's the same as before.
Fix Incrementing Sequence Number For Retransmitted DTMF End Packets
In Asterisk 1.4+, a fix was put in place to increment the sequence number for
retransmitted DTMF end packets. With the introduction of the RTP engine API in
1.8, the sequence number was no longer being incremented. This patch fixes this
regression as well as cleans up a few lines that were not doing anything.
(closes issue ASTERISK-20295)
Reported by: Nitesh Bansal
Tested by: Michael L. Young
Patches:
01_rtp_event_seq_num.patch uploaded by Nitesh Bansal (license 6418)
asterisk-20295-dtmf-fix-cleanup.diff uploaded by Michael L. Young (license 5026)
Fix memory leak when CEL is successfully written to PostgreSQL database
PQClear is not called when the result object of a call to PQExec has a
status of PGRES_COMMAND_OK. Interestingly enough, the off nominal case was
handled properly, so this memory leak only occurred when CEL records were
successfully written.
This patch properly clears the result in the nominal code path.
Prevent crash on shutdown due to refcount error on queues container.
When app_queue is unloaded, the queues container has its refcount
decremented, potentially to 0. Then the taskprocessor responsible
for handling device state changes is unreferenced. If the
taskprocessor happens to be just about to run its task, then it
will create and destroy an iterator on the queues container.
This can cause the refcount on the queues container to increase to
1 and then back to 0. Going back to 0 a second time results in
double frees.
This failure was seen periodically in the testsuite when Asterisk
would shut down.
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Merged revisions 372089 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Help prevent ringing queue members from being rung when ringinuse set to no.
Queue member status would not always get updated properly when the member
was called, thus resulting in the member getting multiple calls. With this
change, we update the member's status at the time of calling, and we also
check to make sure the member is still available to take the call before
placing an outbound call.
(closes issue ASTERISK-16115)
reported by nik600
Patches:
app_queue.c-svn-r370418.patch uploaded by Italo Rossi (license #6409)
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Merged revisions 372048 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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AST-2012-012: Resolve AMI User Unauthorized Shell Access through ExternalIVR
The AMI Originate action can allow a remote user to specify information that can
be used to execute shell commands on the system hosting Asterisk. This can
result in an unwanted escalation of permissions, as the Originate action, which
requires the "originate" class authorization, can be used to perform actions
that would typically require the "system" class authorization. Previous attempts
to prevent this permission escalation (AST-2011-006, AST-2012-004) have sought
to do so by inspecting the names of applications and functions passed in with
the Originate action and, if those applications/functions matched a predefined
set of values, rejecting the command if the user lacked the "system" class
authorization. As reported by IBM X-Force Research, the "ExternalIVR"
application is not listed in the predefined set of values. The solution for
this particular vulnerability is to include the "ExternalIVR" application in the
set of defined applications/functions that require "system" class authorization.
Unfortunately, the approach of inspecting fields in the Originate action against
known applications/functions has a significant flaw. The predefined set of
values can be bypassed by creative use of the Originate action or by certain
dialplan configurations, which is beyond the ability of Asterisk to analyze at
run-time. Attempting to work around these scenarios would result in severely
restricting the applications or functions and prevent their usage for legitimate
means. As such, any additional security vulnerabilities, where an
application/function that would normally require the "system" class
authorization can be executed by users with the "originate" class authorization,
will not be addressed. Instead, the README-SERIOUSLY.bestpractices.txt file has
been updated to reflect that the AMI Originate action can result in commands
requiring the "system" class authorization to be executed. Proper system
configuration can limit the impact of such scenarios.
(closes issue ASTERISK-20132)
Reported by: Zubair Ashraf of IBM X-Force Research
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Merged revisions 371998 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r372020 | mjordan | 2012-08-30 11:22:54 -0500 (Thu, 30 Aug 2012) | 17 lines
AST-2012-013: Resolve ACL rules being ignored during calls by some IAX2 peers
When an IAX2 call is made using the credentials of a peer defined in a dynamic
Asterisk Realtime Architecture (ARA) backend, the ACL rules for that peer are
not applied to the call attempt. This allows for a remote attacker who is aware
of a peer's credentials to bypass the ACL rules set for that peer.
This patch ensures that the ACLs are applied for all peers, regardless of their
storage mechanism.
(closes issue ASTERISK-20186)
Reported by: Alan Frisch
Tested by: mjordan, Alan Frisch
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Merged revisions 372015 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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In r294740, the CODING-GUIDELINES was removed from the doc folder in favor
of the content on the Asterisk wiki. Some folks still look in the doc folder
initially for coding guideline suggestions; as such, this patch adds a
CODING-GUIDELINES file back into the doc folder. The content of the file
merely points to the correct page on the Asterisk wiki where the coding
guidelines currently live.
(closes issue ASTERISK-20279)
Reported by: Andrew Latham
Patches:
CODING-GUIDELINES.diff uploaded by Andrew Latham (license 5985)
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Merged revisions 371961 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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The v1.8 -r369258 change to fix the F and F(x) action logic introduced a
regression in passing the hangup cause from the called channel to the
caller channel.
(closes issue ASTERISK-20287)
Reported by: Konstantin Suvorov
Patches:
app_dial_hangupcause.patch (license #6421) patch uploaded by Konstantin Suvorov (modified)
Tested by: rmudgett
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Merged revisions 371860 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Fix incorrect documentation of the MailboxStatus manager command.
The "Waiting" field was misdocumented as reporting the number of
messages waiting. In reality, it simply indicated the presence or
absence of waiting messages.
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Merged revisions 371782 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r371789 | mmichelson | 2012-08-27 16:49:51 -0500 (Mon, 27 Aug 2012) | 13 lines
Fix misleading documentation in agents.conf.sample regarding ackcall usage.
The documentation made it sound as if the DTMF acknowledgment was needed
at the time the agent logs in, rather than when the agent is called. This
is likely a relic from the days when there were multiple ways of logging
in agents.
(closes issue AST-962)
reported by Steve Pitts
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Merged revisions 371787 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Fixes ast_rwlock_timed[rd|wr]lock for BSD and variants.
The original implementations simply wrap pthread functions, which take
absolute time as an argument. The spinlock version for systems without
those functions treated the argument as a delta. This patch fixes the
spinlock version to be consistent with the pthread version.
When compiling with BETTER_BACKTRACES enabled, Asterisk will sometimes
crash when "core show locks" is run. This happens regularly in the
testsuite since several tests run "core show locks" to help with
debugging. This seems to be a fault with libraries on certain operating
systems (notably CentOS 6.2/6.3) running on virtual machines and
utilizing gcc 4.4.6.
(closes issue ASTERISK-20090)
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Merged revisions 371690 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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In some cases, recovering lost packets using the secondary packet
recovery mechanism with UDPTL/T.38 can result in the recovery of
zero-length packets. These must be ignored or the frame generated from
them can cause segfaults and allocation failures.
(closes issue ASTERISK-19762)
(closes issue ASTERISK-19373) Reported-by: Benjamin (bulkorok) Reported-by: Rob Gagnon (rgagnon)
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Merged revisions 371544 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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When formatting documentation fields, the XML documentation parser calls
xmldoc_get_formatted. This function allocates a string buffer at the
beginning of its routine. Unfortunately, on certain code paths, it also
calls xmldoc_string_cleanup, which assumes that it will create the string
buffer. The previously allocated string buffer is then leaked by the
xmldoc_string_cleanup routine.
Now: we don't do that.
(closes issue AST-932)
Reported by: Alexander Homig
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Merged revisions 371469 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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When Asterisk is built with TEST_FRAMEWORK defined, Asterisk will now
generate TestEvent AMI events on subsystem reloads such as cdr, dnsmgr,
extconfig, etc.
(issue PQ-1126)
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Merged revisions 371436 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Add module reload instrumentation for TEST_FRAMEWORK
This adds AMI events for module reloads when Asterisk is built with
TEST_FRAMEWORK enabled and corrects generation of the module load AMI
event.
(issue PQ-1126)
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Merged revisions 371393 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r371398 | twilson | 2012-08-16 17:50:12 -0500 (Thu, 16 Aug 2012) | 13 lines
Handle integer over/under-flow in ast_parse_args
The strtol family of functions will return *_MIN/*_MAX on overflow. To
detect when an overflow has happened, errno must be set to 0 before
calling the function, then checked afterward.
(closes issue ASTERISK-20120)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2073/
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Merged revisions 371392 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK
Under certain conditions, a SIP transaction involving directmedia wouldn't
trigger a re-invite because the SDP answer was included in an ACK instead
of in a message that we would have triggered the invite with. This patch
just queues a source change control frame if the dialog is using
directmedia when we find sdp for an ACK.
(closes issue AST-913)
Reported by: Thomas Arimont
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Merged revisions 371337 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Fix bug where final queue member would not be removed from memory.
If a static queue had realtime members, then there could be a potential
for those realtime members not to be properly deleted from memory.
If the queue's members were loaded from realtime and then all the
members were deleted from the backend, then the queue would still
think these members existed. The reason was that there was a short-
circuit in code such that if there were no members found in the
backend, then the queue would not be updated to reflect this.
Note that this only affected static queues with realtime members.
Realtime queues with realtime members were unaffected by this issue.
(closes issue ASTERISK-19793)
reported by Marcus Haas
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Merged revisions 371306 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Avoid unconditional NULLing of mwipvt on relatedpeer on SIP dialog destruction
The other instance of this bug was fixed by jcolp/file in r121496. If
we are destroying a dialog only set the MWI dialog pointer on the
related peer to NULL if it is the dialog currently being destroyed.
The helper function, get_address_family_filter, in chan_sip for dns resolution
by address family was not recognizing the websockets transport and resulting in
a null pointer being sent to functions in netsock2, in an attempt to determine
if we are bound to ANY address ([::]) or not.
This patch fixes this issue by handling the transport types SIP_TRANSPORT_WS and
SIP_TRANSPORT_WSS which results in a sock address being set properly for use in
determining the address family.
(closes issue ASTERISK-20221)
Reported by: Sven Beisiegel
Tested by: Sven Beisiegel, James Mortensen
Patches:
asterisk-20221-ws-family-filter.diff uploaded by Michael L. Young (license 5026)
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r371251 | elguero | 2012-08-14 20:43:23 -0500 (Tue, 14 Aug 2012) | 4 lines
Reverting this change that was meant for branch 11.
Fix problem where incorrect pointer was checked for nullity.
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Merged revisions 371198 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r371203 | kmoore | 2012-08-13 15:04:15 -0500 (Mon, 13 Aug 2012) | 13 lines
Add test instrumentation
This adds test instrumentation for loading and unloading of modules
and for certain actions in MeetMe to be used in the testsuite or any
other consumer of AMI events. These will only be generated when
Asterisk is built with TEST_FRAMEWORK enabled.
(issue PQ-1131)
(issue PQ-1133)
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Merged revisions 371201 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Fix a couple of documentation problems in app_queue.c
* The RemoveQueueMember app made mention of options that could
be passed in, but no options are supported. I have removed the
listing of options from the documentation.
* The RQMSTATUS variable did not list "NOTDYNAMIC" as a possible
value that could be set.
(closes issue AST-949)
reported by Steve Pitts
(closes issue AST-954)
reported by Steve Pitts
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Merged revisions 371141 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Automerge script [Fri, 10 Aug 2012 17:23:50 +0000 (17:23 +0000)]
Merged revisions 371090 via svnmerge from
file:///srv/subversion/repos/asterisk/branches/10
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r371090 | may | 2012-08-10 11:46:38 -0500 (Fri, 10 Aug 2012) | 12 lines
remove ALREADYGONE flag on ooh323 call data by ooh323_indicate
(CONGESTION/BUSY) due to call hasn't gone there really.
This indication arrive from asterisk core not h.323 stack
The documentation for the x flag for MeetMe incorrectly described its
function as closing down the conference when the last marked user left.
It actually causes the users with that flag to leave the conference
when the last marked user exits. The functionality of this flag is not
changing.
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Merged revisions 370985 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r370989 | may | 2012-08-09 13:05:34 -0500 (Thu, 09 Aug 2012) | 5 lines
change opening h323 logfile with append mode instead of overwrite
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Merged revisions 370988 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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When a channel hangs up while being spied upon and the option to exit the
ChanSpy application when the spied on channel hangs up is set,
ast_autochan_destroy is not being called and therefore a reference to the spied
upon channel is not removed.
The symptom being reported was that when using func_group in the dialplan and
calling "group show channels" at the cli, the spied upon channel was still
being shown while "core show channels" showed that the channel was not up.
This patch calls ast_autochan_destroy when a spied upon channel hangs up and
the option to exit the ChanSpy application is set, removing the reference to
the channel allowing the count for the group that the spied channel was part of
to be decremented.
(closes issue ASTERISK-17515)
Reported by: Arkadiusz Malka
Tested by: Alexandr Gordeev, Michael L. Young
Patches:
asterisk-17515-destroy-autochan.diff
uploaded by Michael L. Young (license 5026)
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Merged revisions 370952 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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AST_CAUSE_NOTDEFINED is a placeholder for usage when there is no cause
information. As such, it should not be defined and translatable as a
cause.
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Merged revisions 370923 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Fix error in the "IPorHost" section of a SIP dialstring.
This is based on the review request posted by Walter Doekes
(referenced lower in the commit message)
The main fix here is to treat the IPorHost portion of the dial
string as a temporary outbound proxy. This ensures requests
get sent to the proper location.
Due to the age of the request, some parts were no longer relevant.
For instance, the request moved outbound proxy parsing code into
a single method. This is done in a previous commit, so it was not
necessary to do again.
Also, the review request fixed some errors with regards to request
routing for CANCEL and ACK requests. This has also been fixed in
more recent commits.
(closes issue ASTERISK-19677)
reported by Walter Doekes
Schedule pokes of registered SIP peers within a given timespan after SIP reload
With a large number of SIP peers registered, performing a SIP reload causes a
flood of SIP OPTIONS request packets. These are immediately sent out, and, as
responses come back, can cause peers to be flagged as 'lagged' due to handling
of the many response messages.
This fix prevents this "packet storm" and schedules the pokes for a random
time. That time varies between 1 ms and the peer's qualify time, or, if
the qualify time is unknown, the global qualifyfreq setting.
The committed patch has some very small modifications to the patch schmidts
wrote for the review.
This replaces all calls to alloca() with ast_alloca() which calls gcc's
__builtin_alloca() to avoid BSD semantics and removes all NULL checks
on memory allocated via ast_alloca() and ast_strdupa().
When Asterisk servers are set up back-to-back, and
direct media is to be used betweeen endpoints, it is
fairly common for the two Asterisk servers to send
direct media reinvites to each other simultaneously.
This results in 491s and ACKs being exchanged between
the servers. While the media eventually gets set up
properly, the problem is that there can be a noticeable
delay for the streams to stabilize.
This patch adds a new directmedia option called "outgoing".
With this set, an immediate direct media reinvite will only
be sent if the call direction is outgoing. For incoming
dialogs, an immediate direct media reinvite will not be sent,
but further "reactionary" direct media reinvites may be sent.
For those who are having some deja vu, that's because this
patch was originally committed to trunk since there is a
new configuration option added. After seeing a bug report
about audio being slow to set up on SIP calls, it became
apparent that this patch would be the best solution for
resolving the issue. The patch is unintrusive and will
have no effect unless the option is explicitly enabled.
(closes issue AST-896)
reported by Thomas Arimont
(closes issue ASTERISK-19857)
reported by Matt Jordan
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Merged revisions 370618 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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res_agi: Add message indicating need for \n character in verbose message
The while loop responsible for reading AGI messages from a fastAGI service
can end up looping indefinitely when an AGI script fails to indicate the end
of a message with a \n character. This patch adds an indication that we are
expecting a \n character to end the message to make it more clear to users
that this is necessary if they are receiving this warning over and over.
Terry Wilson [Wed, 25 Jul 2012 03:45:20 +0000 (03:45 +0000)]
Revert a change that broke compilation
1) There is no such function as ast_ref()
2) The patch was originally credited as the one uploaded by Guenther
Kelleter (license 6372) via issue AST-921, but the patch committed
was not the patch referenced on the issue.
3) Guenther Kelleter's patch was actually correct. It moved the
ast_free above the presencechange_cleanup label. I am not
committing his change as it is not technically necesary--calling
ast_free(NULL) is perfectly safe and I worry that moving the
ast_free outside of the label could lead to future bugs if
someone ever adds another failure conditional and expects
'goto presencechange_cleanup;' to clean up after everything.
Revision 370205 added the use of a datastore attached to a dummy channel to
resolve a memory leak, but ast_dummy_channel_destructor() in this branch did
not free datastores, resulting in a continued (but slightly smaller) memory
leak. This patch backports the change to free said datastores from the Asterisk
trunk.
(related to issue AST-916)
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Merged revisions 370360 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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The current implementation of RFC 2833 DTMF handling in res_rtp_asterisk will,
if a packet arrives out of order, drop the packet. This is to prevent
duplicate ton generation in the Asterisk core. Since the RTP layer does not
buffer data itself, this is the only option the RTP layer currently has for
handling packets that arrive out of order.
For the most part, this doesn't matter. For a particular digit, so long as a
BEGIN packet arrives before the first END packet, the digit will be produced.
If subsequent BEGIN packets arrive interleaved with the ENDs, they will be
dropped; likewise, if the BEGIN or END packets themselves are out of order,
those packets are dropped but sufficient information is conveyed to the
Asterisk core to produce the appropriate digit.
For certain sequences of DTMF packets - most notably when, for a particular
digit, an END packet arrives before any BEGIN packet for that digit - this
is a real problem. When an END arrives before any BEGINs, the END packet is
dropped - but at the same time, it causes subsequent BEGIN packets for that
digit to be ignored. When the next in order END packet arrives, it too is
dropped - Asterisk believes that there was no initial BEGIN.
The solution this patch provides is to trust the END packet to convey the
information needed for the Asterisk core to produce the DTMF digit. If we
receive an END packet, and it:
* Has a timestamp greater then the last timestamp received from an END
packet
* Does not have the same sequence number as the last received sequence
number (and is thus not an END packet retransmission)
Then we send the END frame up to the Asterisk core. It contains enough
DTMF information for Asterisk to produce the digit.
On the other hand, if we receive a BEGIN or continuation packet that occurs
with a timestamp equal to or less then the last END timestamp, then we've
received something out of order - but we already have received enough
information to produce the digit. These packets are dropped.
Much thanks goes to Olle Johansson (oej) for providing the idea for this
solution.
Review: https://reviewboard.asterisk.org/r/2033/
(issue ASTERISK-18404)
Reporter: Stephane Chazelas
Tested by: Matt Jordan
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Fix compilation error when MALLOC_DEBUG is enabled
To fix a memory leak in CEL, a channel datastore was introduced whose
destruction function pointer was pointed to the ast_free macro. Without
MALLOC_DEBUG enabled this compiles as fine, as ast_free is defined as free.
With MALLOC_DEBUG enabled, however, ast_free takes on a definition from a
different place then utils.h, and became undefined. This patch resolves this
by using a reference to ast_free_ptr. When MALLOC_DEBUG is enabled, this
calls ast_free; when MALLOC_DEBUG is not enabled, this is defined to be
ast_free, which is defined to be free.
(issue AST-916)
Reported by: Thomas Arimont
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Resolve severe memory leak in CEL logging modules.
A customer reported a significant memory leak using Asterisk 1.8. They
have tracked it down to ast_cel_fabricate_channel_from_event() in
main/cel.c, which is called by both in-tree CEL logging modules
(cel_custom.c and cel_sqlite3_custom.c) for each and every CEL event
that they log.
The cause was an incorrect assumption about how data attached to an
ast_channel would be handled when the channel is destroyed; the data
is now stored in a datastore attached to the channel, which is
destroyed along with the channel at the proper time.
Ensure that all ast_datastore_info structures are 'const'.
While addressing a bug, I came across a instance of 'struct ast_datastore_info'
that was not declared 'const'. Since the API already expects them to be
'const', this patch changes the declarations of all existing instances
that were not already declared that way.
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Merged revisions 370183 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Prevent double uri_escaping in chan_sip when pedantic is enabled
If pedantic mode is enabled, outbound invites will have double-escaped
contacts. This avoids setting an already-escaped string into a field
where it is expected to be unescaped.
(closes issue ASTERISK-20023) Reported-by: Walter Doekes
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Merged revisions 369993 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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The documentation for DEC in func_math.c was incorrect. Looks like a copy and
paste error.
(Closes issue ASTERISK-20095)
Reported by: Billy Chia
Tested by: Michael L. Young
Patches:
func_math.patch uploaded by Billy Chia (license 6381)
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Merged revisions 369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Allow the REALTIME() function to report errors back to the caller.
Also, do more error checking on the arguments specified to the REALTIME()
function and clarify the documentation. While I was editing the file, a
few coding guidelines fixups, as well.
chan_sip: Fix small behavioral change accidentally introduced in r369750
When removing the warning for AST_CONTROL_FLASH from sip_indicate, I also
inadvertently changed the return value, which would likely make the indication
not be sent in audio. This fixes that while still removing the warning message.
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Merged revisions 369792 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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chan_sip: Add case for FLASH control frames so that we don't display a warning.
chan_sip channels can receive flash control frames when connected to analog
phones and possibly for other reasons. There really isn't a reason to warn when
these frames are received, we can safely ignore them.
Patches:
dahdi_sip_flash.diff uploaded by Jonathan Rose (license 6182)
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Merged revisions 369750 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Remove a superfluous and dangerous freeing of an SSL_CTX.
The problem here is that multiple server sessions share
a SSL_CTX. When one session ended, the SSL_CTX would be
freed and set NULL, leaving the other sessions unable to
function.
The code being removed is superfluous because the SSL_CTX
structures for servers will be properly freed when ast_ssl_teardown
is called.
(closes issue ASTERISK-20074)
Reported by Trevor Helmsley
Patches:
ASTERISK-20074.diff uploaded by Mark Michelson (license #5049)
Testers:
Trevor Helmsley
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Merged revisions 369731 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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The bridge thread was exiting but was never being
reaped using pthread_join(). This has been fixed now
by calling pthread_join() in ast_bridge_destroy().
(closes issue ASTERISK-19834)
Reported by Marcus Hunger
The heard and deleted arrays in the voicemail state structure were not
handled properly following the memory leak fix in r354890 and a fix for
an invalid free in r356797. This could result in accessing and writing
into freed memory. The allocation for these arrays has been reworked
to avoid the possibility of invalid frees, access of freed memory, and
crashes that were occurring as a result of this.
Locking around accesses and modifications of the voicemail state
structure members dh_arraysize, heard, and deleted has been added to
prevent simultaneous modification and access when IMAP storage is in
use. If IMAP storage is not in use, this locking is not compiled in.
projects / thirdparty / asterisk.git / log
